1. Field of the Invention
The present invention relates to a 1-bit signal processor.
2. Description of the Prior Art
It has been proposed to process 1-bit signals with Delta Sigma Modulators (DSM). The 1-bit signals may be audio signals and the invention is described herein by way of example with reference to audio signals.
Background to the present invention will now be described by way of example with reference to FIGS. 1, 2 and 3 of the accompanying drawings of which FIG. 1 is a block diagram of a known Delta-Sigma Modulator, FIG. 2 is a block diagram of a previously proposed Delta-Sigma Modulator configured as an nth order filter section and FIG. 3 shows a noise shaping characteristic.
It is known to convert an analogue signal to a digital form by sampling the analogue signal at at least the Nyquist rate and encoding the amplitudes of the samples by an m bit number. Thus if m=8, the sample is said to be quantized to an accuracy of 8 bits. In general m can be any number of bits equal to or greater than 1.
For the purpose of quantizing to only 1 bit, it is known to provide an analogue to digital converter (ADC) known either as a xe2x80x9cSigma-Delta ADCxe2x80x9d or as a xe2x80x9cDelta-Sigma ADCxe2x80x9d. Herein the term xe2x80x9cDelta-Sigmaxe2x80x9d is used. Such an ADC is described in for example xe2x80x9cA Simple Approach to Digital Signal Processingxe2x80x9d by Craig Marven and Gillian Ewers ISBN 0-904.047-00-8 published 1993 by Texas Instruments.
Referring to FIG. 1 in an example of such an ADC, the difference 1 (Delta) between an analogue input signal and the integral 2 (Sigma) of the 1-bit output signal is fed to a 1-bit quantizer 3. The output signal comprises bits of logical value 0 and 1 but representing actual values of xe2x88x921 and +1 respectively. The integrator 3 accumulates the 1-bit outputs so that value stored in it tends to follow the value of the analog signal. The quantizer 3 increases (+1) or reduces (xe2x88x921) the accumulated value by 1-bit as each bit is produced. The ADC requires a very high sampling rate to allow the production of an output bit stream the accumulated value of which follows the analogue signal.
The term xe2x80x9c1-bitxe2x80x9d signal as used in the following description and in the claims means a signal quantized to an accuracy of 1 digital bit such as is produced by a Delta-Sigma ADC.
A Delta-Sigma Modulator (DSM) configured as nth order filter section for directly processing a 1-bit signal was proposed by N. M. Casey and James A. S. Angus in a paper presented at 95th AES Convention Oct. 7-10, 1993 New York, USA entitled xe2x80x9cOne Bit Digital Processing of Audio Signalsxe2x80x9dxe2x80x94Signal Processing: Audio Research Group, The Electronics Department, The University of York, Heslington, York YO1 5DD England. FIG. 2 shows a 3rd order (n=3) version of such a DSM filter section.
Referring to FIG. 2, the DSM has an input 4 for a 1-bit audio signal and an output 5 at which a processed a 1-bit signal is produced. The bits of the 1-bit signal are clocked through the DSM by known clocking arrangements which are not shown. The output 1-bit signal is produced by a 1-bit quantizer Q which is for example a comparator having a threshold level of zero. The DSM has three stages each comprising a first 1-bit multiplier a1, a2, a3 connected to the input 4, a second 1-bit multiplier c1, c2, C3 connected to the output 5, an adder 61, 62, 63 and an integrator 71, 72, 73.
The 1-bit multipliers multiply the received 1-bit signal by p bit coefficients A1, A2, A3, C1 C2, C3 producing p bit products which are added by the adders 61, 62, 63 and the sums passed to the integrators 7. In the intermediate stages the adders 62, 63 also sum the output of the integrator of the preceding stage. A final stage comprises another 1-bit multiplier A4 connected to the input which multiplies the input signal by a p bit coefficient A4 and an adder 64 which adds the product to the output of the integrator 73 of the preceding stage. The sum is passed to the quantizer Q.
Within the DSM, two""s complement arithmetic is used to represent the positive and negative p bit numbers. The input to the quantizer Q may be positive, quantized at the output as +1 (logical 1) or negative quantized at the output as xe2x88x921 (logical 0).
As observed by Casey and Angus xe2x80x9ca one bit processor . . . will produce a one bit output that contains an audio signal that is obscured by noise to an unacceptable level and it is imperative the quantization noise is suitably shapedxe2x80x9d. The noise which obscures the audio signal is the quantization noise produced by the quantizer Q.
The quantizer Q may be modelled as an adder which has a first input receiving an audio signal and a second input receiving a random bit stream (the quantization noise) substantially uncorrelated with the audio signal. Modelled on that basis, the audio signal received at the input 4 is fed forward by multipliers a1, a2, a3, a4 to the output 5 and fed back by multipliers c1, c2, c3 from the output 5. Thus coefficients A1 to A4 in the feed forward path define zeros of the Z-transform transfer function of the audio signal and coefficients C1-C3 in the feed back path define poles of the transfer function of the audio signal.
The noise signal is fed-back from the quantizer by the multipliers C1-C3 so that coefficients C1-C3 define poles of the transfer function of the noise signal. The transfer function of the noise signal is not the same as the transfer function of the input signal.
The coefficients A1 to A4 and C1 to C3 are chosen to provide circuit stability amongst other desired properties.
The coefficients C1-C3 are chosen to provide noise shaping so as to minimise quantization noise in the audio band, as shown for example in FIG. 3 by the full line 31.
The coefficients A1-A4 and C1-C3 are also chosen for a desired audio signal processing characteristic.
The coefficients A1-A4 and C1-C3 may be chosen by:
a) finding the Z-transform H(z) of the desired filter characteristicxe2x80x94e.g noise shaping function; and
b) transforming H(z) to coefficients.
This may be done by the methods described in the paper
xe2x80x9cTheory and Practical Implementation of a Fifth Order Sigma-Delta A/D Converter, Journal of Audio Engineering Society, Volume 39, no. 7/8, 1991 July/August by R. W Adams et al.xe2x80x9d
and in
the paper by Casey and Angus mentioned herein above using the knowledge of these skilled in the art. One way of calculating the coefficients is outlined in the accompanying Annex A.
The various papers mentioned herein above consider only nth order filter sections.
For high quality audio recording it is conventional to use a microphone having a differential or double-ended analogue output. Proposed 1-bit signal processors including DSMs require non-differential or single-ended signals. One proposal for converting differential signals to non-differential signals is to add using an analogue adder the analogue differential signals output by the microphone and then converting the resultant non-differential signal to 1-bit digital form.
According to the present invention there is provided a signal processor having a pair of inputs for receiving first and second analogue signals which are a differential pair of signals,
a pair of 1-bit analogue to digital converters coupled to the respective inputs to convert the first and second signals to 1-bit digital form, and
a Delta Sigma Modulator having a pair of inputs coupled to receive the respective first and second 1-bit signals for combining the differential pair to form a non-differential 1-bit signal.
In a preferred embodiment of the invention the pair of input signals are produced by an audio signal source such as a differential microphone. By converting the two differential analogue outputs of the microphone to corresponding 1-bit signals and then combining the two 1-bit signals in a DSM, the signal to noise ratio is improved (compared to the prior proposal discussed above). The improvement arises because both the noise in the differential analogue signals and the quantization noise produced in the analogue to digital converters is uncorrelated and so both types of noise are reduced relative to the signal in the DSM combiner.
In a preferred embodiment of the present invention, the Delta Sigma Modulator comprises an nth order (where n is greater than or equal to 1) Delta Sigma Modulator having a first input for receiving a first 1-bit signal, a second input for receiving a second 1-bit signal, a quantizer for requantizing a p bit signal to 1-bit form the requantized signal being the output signal of the processor, a plurality of signal combiners including a first combiner for forming an integral of an additive combination of the product of the first signal and a first coefficient and of the product of the second signal and a second coefficient and of the product of the output signal and a third coefficient, at least one intermediate combiner for forming an integral of an additive combination of the product of the first signal and a first coefficient and of the product of the second signal and a second coefficient and of the product of the output signal and a third coefficient and of the integral of the preceding stage, and a final combiner for forming an additive combination of the product of the first signal and a first coefficient and of the product of the second signal and a second coefficient and of the integral of the preceding stage to form the said p bit signal which is requantized by the quantizer.
Thus the DSM combines the first and second signals. Coefficients multiplies of the said combiners operate on 1-bit signals and so coefficient multiplication is performed as 1-bit multiplication avoiding the need for p bit multipliers which are uneconomic.
Furthermore the DSM also provides noise shaping.
The said first and second coefficients applied to the first and second signals may be fixed in which case the DSM acts as an adder which adds the first and second signals in fixed proportions defined by the said coefficients.
The said first and second coefficients applied to the first and second signals may be variable in which case the DSM acts as a mixer and/or fader.
The first and second coefficients define zeroes of the input signal transfer function and may be fixed or variable, but the third coefficients define poles of the input signal transfer function and are fixed.